Sound dropping out after conference or hold

Dear forum users,

We are having the following issue:

Case1:
When someone call us on our external number, and colleague1 picks up the phone is all going perfect. But when colleague1 doesn’t know the correct answer and puts the customer on hold the customer gets music as planned. Now colleague1 calls internally with colleague2. When colleague1 hangs up on colleague2 and picks up the customer again, he can hear the customer but the customer cannot hear colleague1.

Case2:
colleague1 is talking with colleague2, now colleague3 calls colleague1 he picks him up and put the phone in conference so colleague1, colleague2 and colleague3 can talk and speak to each other. But now the sound dropped for colleague2 or colleague3 (thats random)

The sound dropout is not always its like 50/60% of all conferences, so its hard to troubleshoot.

Our setup:
We have 1 VoIP server in our datacenter. Everyone has a SNOM360 or SNOM370 VoIP Phone on his desk. We are all working from our homes, so we do not have a office with people in it. We are all using different Internet providers.
We have a voip connection from Matrix and MyDivert.

What did i try so far:

  • I have changed some settings with NAT to on/off
  • Switched all codecs to: alaw
  • Our incoming connection from Matrix and MyDivert are also alaw
  • used a stunserver on the phones at home.

Strange thing:
Before we used FreePBX we had a old Asterisk server without a web gui. When we used that everything was working smoothly, beside when u wanted to change voicemail or menu sound it takes u like 10 times more time then with FreePBX.

So hopefully someone can help me/push me in the correct way, like i said the sound drop out (one way) is not always but like 6 out of 10 phone calls tested on the same person.

Kind regards,

Kevin Jansen

We would have to see your call logs when it dropped out. Anything else is speculation

Dear,

Hard to reproduce, since when i am testing it it goes fine:) But lucky i managed to reproduce it and hope i have the correct log for you:

At this part when phone1 transfers his current conversation to phone 2 the sound dropped:

[2014-10-06 17:53:16] DEBUG[1708][C-0000d282] sip/sdp_crypto.c: Accepting crypto tag 1
[2014-10-06 17:53:16] DEBUG[1708][C-0000d282] sip/sdp_crypto.c: Crypto line: a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:bpb6i+SY6ndAluhTY2vETqrvmhumldvRov6KRkRb
[2014-10-06 17:53:16] WARNING[1708][C-0000d282] chan_sip.c: Declining non-primary audio stream: audio 53522 RTP/AVP 9 0 8 3 99 108 18 101
[2014-10-06 17:53:16] DEBUG[1708][C-0000d282] sip/sdp_crypto.c: Crypto line: a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:bpb6i+SY6ndAluhTY2vETqrvmhumldvRov6KRkRb
[2014-10-06 17:53:16] VERBOSE[16838][C-0000d282] res_musiconhold.c: – Started music on hold, class ‘default’, on SIP/1337-00009e0b
[2014-10-06 17:53:18] VERBOSE[1708][C-0000d27e] res_musiconhold.c: – Stopped music on hold on SIP/Matrix-00009e0a
[2014-10-06 17:53:18] VERBOSE[1708][C-0000d27e] res_musiconhold.c: – Stopped music on hold on SIP/1337-00009e0b
[2014-10-06 17:53:18] VERBOSE[16781][C-0000d280] pbx.c: – Executing [h@macro-dialout-trunk:1] Macro(“SIP/1048-00009e08”, “hangupcall,”) in new stack
[2014-10-06 17:53:18] VERBOSE[16781][C-0000d280] pbx.c: – Executing [s@macro-hangupcall:1] GotoIf(“SIP/1048-00009e08”, “1?theend”) in new stack
[2014-10-06 17:53:18] VERBOSE[16781][C-0000d280] pbx.c: – Goto (macro-hangupcall,s,3)
[2014-10-06 17:53:18] VERBOSE[16781][C-0000d280] pbx.c: – Executing [s@macro-hangupcall:3] ExecIf(“SIP/1048-00009e08”, “0?Set(CDR(recordingfile)=)”) in new stack
[2014-10-06 17:53:18] VERBOSE[16781][C-0000d280] pbx.c: – Executing [s@macro-hangupcall:4] Hangup(“SIP/1048-00009e08”, “”) in new stack
[2014-10-06 17:53:18] VERBOSE[16781][C-0000d280] app_macro.c: == Spawn extension (macro-hangupcall, s, 4) exited non-zero on ‘SIP/1048-00009e08’ in macro ‘hangupcall’
[2014-10-06 17:53:18] VERBOSE[16781][C-0000d280] pbx.c: == Spawn extension (macro-dialout-trunk, h, 1) exited non-zero on ‘SIP/1048-00009e08’
[2014-10-06 17:53:18] VERBOSE[16781][C-0000d280] app_macro.c: == Spawn extension (macro-dialout-trunk, s, 22) exited non-zero on ‘SIP/1048-00009e08’ in macro ‘dialout-trunk’
[2014-10-06 17:53:18] VERBOSE[16781][C-0000d280] pbx.c: == Spawn extension (from-internal, 0628981040, 5) exited non-zero on ‘SIP/1048-00009e08’

When i check the log for warnings that comes on my screen i see this 3:

[2014-10-06 17:28:55] WARNING[1708] chan_sip.c: Retransmission timeout reached on transmission 543272098a17-r85tmn66klq6 for seqno 133 (Critical Request) – See wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
[2014-10-06 17:28:55] WARNING[1708][C-0000d261] chan_sip.c: Declining non-primary audio stream: audio 50708 RTP/AVP 9 0 8 3 99 108 18 101
[2014-10-06 17:51:18] WARNING[1708] chan_sip.c: Retransmission timeout reached on transmission 543272099d6b-zrl2wbbhg0tn for seqno 130 (Critical Request) – See wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6400ms with no response

Hope u can see/do something with this.

Kind regards,

Kevin Jansen

You are having NAT issues. Specifically RTP timeout, there is no response the critical RTP packet and thus Asterisk hangs up.

Dear,

But is this NAT issue client side or server side?

Kind regards,

Kevin Jansen

There is no way to tell unless we saw your state log from your firewall and analyzed it. Usually it’s client.